Re: [sipx-users] External calls cannot be transferred to voice mail (sipXecs 4.4.0)
Hi Tony, I really appreciate that you took the time to elaborate in detail below. I shall follow-up and perform your suggestions when time permits. Please also see my response below. Best regards, Henry Kwan From: Tony Graziano To: Joegen Baclor Cc: Discussion list for users of sipXecs software ; Henry Kwan Sent: Friday, October 19, 2012 2:36:25 AM Subject: Re: [sipx-users] External calls cannot be transferred to voice mail (sipXecs 4.4.0) Primus is also LINGO. Primus is a large aggregator and also runs a residential service (Lingo). The Lingo service does not support invite without sdp, unless the reinvite is to one of their services and typically only from one of their ATA's. >> OK, I'll ask Primus about this. I think you would do well to ask them if they support this AND it is very important to make sure the invite for the incoming call comes to your server on port 5080. >> I've confirmed with Primus that they could accept signalling on 5060 on >> their side and sent signalling to us on 5080. I don't think your issue is unusual and usually stems from one of 3 core misconfiguration types: 1. Incompatible ITSP - Does a transfer from the AA to a user work? Does a call from a user to another user work? (both as inbound calls via the trunk). Is the original invite coming on port 5080. >> Did not test a transfer from the AA to a user, will try that. >> Call from a user (internal phone) to another user through local dialing plan >> (i.e. 9-...) worked. >> I think the original invite must come on port 5080 as that was the port that >> was forwarded. 5060 was not forwarded. 2. Does the phone ring? If so, how was it configured (manually of by sipx)? Please tell me you didn't register the line manually using the sipx ip address. DNS is VERY important for the refer to voicemail. IF you registered by IP, make sure you add the IP as a domain alias, but really you should NEVER register by IP and expect all things to work well. >> Yes, the phone rang. The phone, Linksys SPA942, was configured via the sipX >> web pages. I then reset the phone and have the configuration downloaded to >> the phone via TFTP (I think this is the mechanism). >> I re-installed sipXecs 4.4 a number of times. Sometimes IP as a domain >> alias would appear automatically. I've also manually done that. In any >> event, that did not help. >> I also run the tests on the configuration test page and everything passed, >> including DNS checks. I've also downloaded Flight Test (I think that was >> the name) and everything passed. 3. Firewall - Unless your firewall is doing 1:1 NAT, you need to make sure it can do Manual AON (static port nat). With pfsense this is easy, but YOU CANNOT create port forward rules until you do this for SIPX becuase they will follow the original NAT type. I sent you a link of how to do this earlier, its pretty straightforward. You should be able to use the pcap tool in pfsense and have it listen on WAN port 5080 and do a capture and see if the ITSP sends the call in on the right port. If not, it will never work right (no matter what version) and you need to ask them if they support this. >> I did not do 1:1 NAT as I was not sure how to do that properly. I've read >> up on it now and will try that out in the future. For port forwarding, I >> did do Manual AON with static port checked on pfSense. I also needed to >> create rules to pass this traffic. Same thing was done for port range 3 >> to 31000. With this setup on pfSense, I could call in from an external >> phone but still could not transfer to voice mail when no one answered. It >> behaved exactly the same as using other routers - fast busy when the attempt >> of transfer was made. >> I have not had time to follow the link that you sent me earlier but will >> definitely read up on it. Good luck! On Fri, Oct 19, 2012 at 1:50 AM, Joegen Baclor wrote: > Transferring ITSP originated calls requires that your ITSP supports INVITE > without SDP. Before barking on something on the system, check first if your > ITSP supports this. If not, there is no way your ITSP will work with sipx > initiated transfers. > > > > On 10/19/2012 01:19 PM, Tony Graziano wrote: >> >> On Fri, Oct 19, 2012 at 1:07 AM, Henry Kwan wrote: >>> >>> My installation was right from the 4.4 ISO. I did try without updating >>> at >>> all but to no avail. >>> >>> My ITSP is Primus Canada. >>> >>> Well I have to admit that I am not knowledgeable in setting up pfSense. >>> In >>> fact I am not knowledgeable on how to produce a pcap or produce a >>> siptrace >>> as Tony suggested. Having said that, I'll continue to play with 4.4 and >>> look into how to perform the tasks suggested when time permits. >>> >> Pfsense >> http://blog.myitdepartment.net/?p=297 >>> >>> In the mean time, 4.2.1 will have to suffice until I can figure out what >>> I >>> did wrong. >>> >>> By the way, my observation regarding the inconsis
Re: [sipx-users] External calls cannot be transferred to voice mail (sipXecs 4.4.0)
Thanks Joegen for your advice, I shall do that. I did not do that prior to taking on this endeavour because I did not know this technical detail. As I stated in my previous posting/email on this subject, I am not an experienced sipXecs user nor SIP knowledgeable. I hope that this forum will tolerate users with low technical subject matter knowledge, like me, to seek advices here such that these learning and experience building processes are fun and rewarding. Should I am coming across as "barking on something on the system", please accept my apology as this was never my intention. I was just trying to ask questions that I could not find answers after searching through various sources on the internet. Again thank you to all generous advices and suggestions. Best regard to all, Henry Kwan From: Joegen Baclor To: Discussion list for users of sipXecs software Cc: Tony Graziano ; Henry Kwan Sent: Thursday, October 18, 2012 11:50:15 PM Subject: Re: [sipx-users] External calls cannot be transferred to voice mail (sipXecs 4.4.0) Transferring ITSP originated calls requires that your ITSP supports INVITE without SDP. Before barking on something on the system, check first if your ITSP supports this. If not, there is no way your ITSP will work with sipx initiated transfers. On 10/19/2012 01:19 PM, Tony Graziano wrote: > On Fri, Oct 19, 2012 at 1:07 AM, Henry Kwan wrote: >> My installation was right from the 4.4 ISO. I did try without updating at >> all but to no avail. >> >> My ITSP is Primus Canada. >> >> Well I have to admit that I am not knowledgeable in setting up pfSense. In >> fact I am not knowledgeable on how to produce a pcap or produce a siptrace >> as Tony suggested. Having said that, I'll continue to play with 4.4 and >> look into how to perform the tasks suggested when time permits. >> > Pfsense > http://blog.myitdepartment.net/?p=297 >> In the mean time, 4.2.1 will have to suffice until I can figure out what I >> did wrong. >> >> By the way, my observation regarding the inconsistent behaviour on restarts >> for both 4.2.1 and 4.4.0 is not an attack on sipXecs. It is just that - an >> observation. Maybe someone can comment if this observation is also only >> experienced by me. If that's the case, I must be a jinx or have a unique >> ability to bring out the worst in sipXecs. >> > I can set up a new system each day and don't experience this behavior. > It's really important to observe how much RAM you have installed (I > prefer 8GB minimum but for very small installs 4GB on 4.4 is enough, > though 8GB should be the minimum for 4.6). >> Best regards to all, >> >> Henry Kwan >> >> >> From: George Niculae >> >> To: Discussion list for users of sipXecs software >> >> Cc: Henry Kwan >> Sent: Thursday, October 18, 2012 6:29:01 PM >> >> Subject: Re: [sipx-users] External calls cannot be transferred to voice mail >> (sipXecs 4.4.0) >> >> On Fri, Oct 19, 2012 at 3:26 AM, Tony Graziano >> wrote: >>> Rather than use an old unsupportable version, produce a pcap from your >>> firewall or produce a siptrace from sipx itself. >>> >>> I don't think your off the cuff observation is exactly right on targetm . >>> Version 4.2 used its own media server while 4.4 uses FreeSWITCH and there >>> are significant close changes. >>> >>> You could also indicate whether or not you followed a tutorial on how to >>> properly configure pfsense and who the itsp is. >>> >> Additionally, if you could try scenario with 4.4 built from ISO, >> without yum updating to latest, and report back, will help identifying >> if issue in latest patches >> >> Thanks >> George >> >> >> >> ___ >> sipx-users mailing list >> sipx-users@list.sipfoundry.org >> List Archive: http://list.sipfoundry.org/archive/sipx-users/ > >___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
[sipx-users] Bug fix release update: sipXecs 4.4.0 update #22
Update #22 : Fri 19. Oct 2012 == - ** No security updates in this update ** - ISO has *not* been rebuilt as decided in release policy. Yum update after installation is recommended for getting these updates. - Thank you all for your continued testing and fixes. Build Log = commit f9c26b7d07384753043385f05def5887732bebd0 Author: dizzy Date: Tue Oct 9 17:27:36 2012 +0300 PT-37419533: Call intro conferences fails with 404 / Not Found commit fe222e6fdc7f16ff04a8d039fe2ccedb6a927cc6 Author: Joegen Baclor Date: Fri Oct 19 18:50:16 2012 +0800 Correcting typo in Message Status body. We are missing an "s" in Message(s)-Waiting. commit bddca93b76c7e8474b48b3b004526084e56d262f Author: Joegen Baclor Date: Fri Oct 19 16:25:47 2012 +0800 PT-38071901: A regression has been introduced after the most recent patches to the issue. Fetching aliases fails if we commit 1d32f6d4301aa08b08a09302cdfa3024d10ee24c Author: George Niculae Date: Thu Oct 18 14:31:49 2012 +0300 PT-37957859 - Allow sending of NIST 100 trying for subscriptions to be configurable - exposed setting in Voicemail MWI services page commit 577b7501cf9ea737475ee722ace786399e0bb224 Author: Joegen Baclor Date: Thu Oct 18 17:17:50 2012 +0800 PT-37957859 - Allow sending of NIST 100 trying for subscriptions to be configurable. for past releases see http://download.sipfoundry.org/pub/sipXecs/ChangeLog-4.4.0 ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] 4.4.0 to 4.4.6
On Fri, Oct 19, 2012 at 10:19 PM, wrote: > Thanks for the info. I will have to read up on GridFS. > Well if you want to query mailbox and voicemail statuses you could use REST API, that's the same no mater if file system or grid fs support George ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] 4.4.0 to 4.4.6
Thanks for the info. I will have to read up on GridFS. - Original Message - From: "George Niculae" To: "Discussion list for users of sipXecs software" Sent: Friday, October 19, 2012 2:58:49 PM Subject: Re: [sipx-users] 4.4.0 to 4.4.6 On Fri, Oct 19, 2012 at 9:51 PM, wrote: > Hi George! > > I yum updated from openuc-stage and verified this now works. > > Just for my own edification, is the mechanism for heard/unheard the same in > OpenUC 4.6 as it was in 4.4 (i.e. ".sta" files), just in mongodb's gridfs? > Same for the from, duration, time received, priority, etc. in ".xml" files? > Hi Andrew, no, it's a different mechansim, ivr supports now mailbox manager plugins and openuc is using one to save / retrieve to / from GridFS. GridFS associates metadata with saved files so all the info like unread, duration, time received are stored in GridFS for openuc (open source project is using the same file system based mailbox manager as in 4.4 and the same mechanism - .sta, .urg xml files) (Thanks for checking the fix) George ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] 4.4.0 to 4.4.6
On Fri, Oct 19, 2012 at 9:51 PM, wrote: > Hi George! > > I yum updated from openuc-stage and verified this now works. > > Just for my own edification, is the mechanism for heard/unheard the same in > OpenUC 4.6 as it was in 4.4 (i.e. ".sta" files), just in mongodb's gridfs? > Same for the from, duration, time received, priority, etc. in ".xml" files? > Hi Andrew, no, it's a different mechansim, ivr supports now mailbox manager plugins and openuc is using one to save / retrieve to / from GridFS. GridFS associates metadata with saved files so all the info like unread, duration, time received are stored in GridFS for openuc (open source project is using the same file system based mailbox manager as in 4.4 and the same mechanism - .sta, .urg xml files) (Thanks for checking the fix) George ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] 4.4.0 to 4.4.6
Hi George! I yum updated from openuc-stage and verified this now works. Just for my own edification, is the mechanism for heard/unheard the same in OpenUC 4.6 as it was in 4.4 (i.e. ".sta" files), just in mongodb's gridfs? Same for the from, duration, time received, priority, etc. in ".xml" files? Thanks! Andy - Original Message - From: "George Niculae" To: "Discussion list for users of sipXecs software" Sent: Friday, October 19, 2012 7:17:52 AM Subject: Re: [sipx-users] 4.4.0 to 4.4.6 On Fri, Oct 19, 2012 at 1:22 PM, George Niculae wrote: > On Fri, Oct 19, 2012 at 4:50 AM, Andrew Pitman > wrote: >> Thanks George! I tried it again today and verified it works. It seems that >> the heard/unheard status of voicemails is not preserved though. All messages >> were "new" after the import. It's no biggie, but is this something that is >> being worked on or something we'll just have to deal with? > > Good catch, let me fix that too. Thanks! > Fix committed, if you're installing from openuc-stage then yum update, if you're installing from openuc then it will take some more time until build published. Either way please let me know if works for you after yum update and test it, I have a button to push :) George ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Getting 483 Too Many Hops Error
You might be able to get the ingate to increase the Max-Forwards in newer firmware. Most of the ingates I have deployed are running firmware at least a year or so old, and i had a ticket open with ingate to add this feature but at the time it wasn't there. Regardless of what the ingate does, you still need good routing. Even if the ingate can hide the low max-forwards and send sipX a new invite with higher max-forwards, sipx will still keep track of the hops as it goes through the system and if it hits zero, it will drop the call. -M >>> Tony Graziano 10/19/12 2:31 PM >>> Except with an ingate we can probably preserve the hop count inside and make it a non issue (I think). On Fri, Oct 19, 2012 at 2:18 PM, Matt White wrote: > Attachments typically don't make ti to the list. Post them to a webserver > and provide a url. > > 483 Too many Hops is generally because the call is getting > referred/re-invited too many times. > > The ITSP will typically send the call with predefined number of "hops" set > in the SIP header. We often find that this changes even within a single > ITSP depending on where the call originates. Most ITSP set this around 10 > hops. > > its not hard for this to decrement down to 0. Call hits the ingate > (1)...Call Goes to an alias on an AA (2), call is sent from the alias to the > AA (3), AA transfers to a hunt group (4)etc. A few hops are used > internally to sipx as well. Eventually it bounces around too many times. > > So for starters see if you can simplify your call routing. > > Its also good to fully qualify call forwarding routes as it removes a hop. > For example. > > Lets say you have extension 201 as a dumy user and it sends calls to hunt > group 202. > When you setup the call forwarding enter it as 202@sipx.domain > > If you just put 202 in their...it will take 2 hops. One when it sends 202 > to the proxy, and a second when the proxy changes 202 to 202@sipx.domain. > You can see it can eat up hops real quick. > > Same goes for alias. If you can route calls to the actual user extension it > will take one less hop than an alias to the same user. > > If all else fails, see if the itsp can up the hops for you. > > -m > Brian Buckles 10/19/12 1:44 PM >>> > > > > > > Sorry if this gets re-posted a second time. I haven't gotten any > confirmation from the admins that this was or wasn't accepted, and this is > something we need to resolve quickly. Please see my previous email below > for details. > Attention SipXecs support community, > We have a client that is running SipXecs 4.4.0. The Session Boarder > Controller is an InGate Siparator and the SIP provider for the SIP trunks is > BroadVox. The client is unable to receive calls to most of their toll free > numbers properly. Our client noticed that prior to calling us, that it > seemed they could get a call to successfully go through from a cell phone to > the toll free numbers, but not from land lines. After further testing it > appears the some calls go through for most cell phone carriers and not > others. It appears most land line calls to the toll free numbers fail, but > there's been one or 2 land based offices that were able to successfully call > the toll free numbers. All calls to the local numbers have went through > without incident as to the best of our knowledge. We talked to the SIP > provider and vendor for the SBC. They found that the SipXecs system is > giving a "483 Too Many Hops" error. I've attached a capture (see the > 726910-3.pcap file ) from the SIP provider as well as capture (See the > "Configured by Ingate" file) from the SBC vendor that was ran on the > SBC. You'll can view the capture from the SBC using a web browser, then > scroll down toward the bottom to see the capture. You can search for the > error code given above. The only changes we are aware of that happened on > the SipXecs system, is that the selt signed cert had expired and was renewed > about a month ago. The issue with the "483 Too Many Hops" has only been > noticed by the client within the last 1.5 to 2 weeks. The toll free number > of 8004144231 was used during the gathering of the 2 attached captures. Can > someone please take a look at the attached capture files ASAP and let us > know of any suggestions you have to resolve this issue? This issue is > having a major impact the client's normal business operations. > > > > ___ > sipx-users mailing list > sipx-users@list.sipfoundry.org > List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ Using or developing for sipXecs from SIPFoundry? Ask me about sipX-CoLab 2013! -- LAN/Tel
Re: [sipx-users] Getting 483 Too Many Hops Error
Except with an ingate we can probably preserve the hop count inside and make it a non issue (I think). On Fri, Oct 19, 2012 at 2:18 PM, Matt White wrote: > Attachments typically don't make ti to the list. Post them to a webserver > and provide a url. > > 483 Too many Hops is generally because the call is getting > referred/re-invited too many times. > > The ITSP will typically send the call with predefined number of "hops" set > in the SIP header. We often find that this changes even within a single > ITSP depending on where the call originates. Most ITSP set this around 10 > hops. > > its not hard for this to decrement down to 0. Call hits the ingate > (1)...Call Goes to an alias on an AA (2), call is sent from the alias to the > AA (3), AA transfers to a hunt group (4)etc. A few hops are used > internally to sipx as well. Eventually it bounces around too many times. > > So for starters see if you can simplify your call routing. > > Its also good to fully qualify call forwarding routes as it removes a hop. > For example. > > Lets say you have extension 201 as a dumy user and it sends calls to hunt > group 202. > When you setup the call forwarding enter it as 202@sipx.domain > > If you just put 202 in their...it will take 2 hops. One when it sends 202 > to the proxy, and a second when the proxy changes 202 to 202@sipx.domain. > You can see it can eat up hops real quick. > > Same goes for alias. If you can route calls to the actual user extension it > will take one less hop than an alias to the same user. > > If all else fails, see if the itsp can up the hops for you. > > -m > Brian Buckles 10/19/12 1:44 PM >>> > > > > > > Sorry if this gets re-posted a second time. I haven't gotten any > confirmation from the admins that this was or wasn't accepted, and this is > something we need to resolve quickly. Please see my previous email below > for details. > Attention SipXecs support community, > We have a client that is running SipXecs 4.4.0. The Session Boarder > Controller is an InGate Siparator and the SIP provider for the SIP trunks is > BroadVox. The client is unable to receive calls to most of their toll free > numbers properly. Our client noticed that prior to calling us, that it > seemed they could get a call to successfully go through from a cell phone to > the toll free numbers, but not from land lines. After further testing it > appears the some calls go through for most cell phone carriers and not > others. It appears most land line calls to the toll free numbers fail, but > there's been one or 2 land based offices that were able to successfully call > the toll free numbers. All calls to the local numbers have went through > without incident as to the best of our knowledge. We talked to the SIP > provider and vendor for the SBC. They found that the SipXecs system is > giving a "483 Too Many Hops" error. I've attached a capture (see the > 726910-3.pcap file ) from the SIP provider as well as capture (See the > "Configured by Ingate" file) from the SBC vendor that was ran on the > SBC. You'll can view the capture from the SBC using a web browser, then > scroll down toward the bottom to see the capture. You can search for the > error code given above. The only changes we are aware of that happened on > the SipXecs system, is that the selt signed cert had expired and was renewed > about a month ago. The issue with the "483 Too Many Hops" has only been > noticed by the client within the last 1.5 to 2 weeks. The toll free number > of 8004144231 was used during the gathering of the 2 attached captures. Can > someone please take a look at the attached capture files ASAP and let us > know of any suggestions you have to resolve this issue? This issue is > having a major impact the client's normal business operations. > > > > ___ > sipx-users mailing list > sipx-users@list.sipfoundry.org > List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ Using or developing for sipXecs from SIPFoundry? Ask me about sipX-CoLab 2013! -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Getting 483 Too Many Hops Error
Attachments typically don't make ti to the list. Post them to a webserver and provide a url. 483 Too many Hops is generally because the call is getting referred/re-invited too many times. The ITSP will typically send the call with predefined number of "hops" set in the SIP header. We often find that this changes even within a single ITSP depending on where the call originates. Most ITSP set this around 10 hops. its not hard for this to decrement down to 0. Call hits the ingate (1)...Call Goes to an alias on an AA (2), call is sent from the alias to the AA (3), AA transfers to a hunt group (4)etc. A few hops are used internally to sipx as well. Eventually it bounces around too many times. So for starters see if you can simplify your call routing. Its also good to fully qualify call forwarding routes as it removes a hop. For example. Lets say you have extension 201 as a dumy user and it sends calls to hunt group 202. When you setup the call forwarding enter it as 202@sipx.domain If you just put 202 in their...it will take 2 hops. One when it sends 202 to the proxy, and a second when the proxy changes 202 to 202@sipx.domain. You can see it can eat up hops real quick. Same goes for alias. If you can route calls to the actual user extension it will take one less hop than an alias to the same user. If all else fails, see if the itsp can up the hops for you. -m >>> Brian Buckles 10/19/12 1:44 PM >>> Sorry if this gets re-posted a second time. I haven't gotten any confirmation from the admins that this was or wasn't accepted, and this is something we need to resolve quickly. Please see my previous email below for details. Attention SipXecs support community, We have a client that is running SipXecs 4.4.0. The Session Boarder Controller is an InGate Siparator and the SIP provider for the SIP trunks is BroadVox. The client is unable to receive calls to most of their toll free numbers properly. Our client noticed that prior to calling us, that it seemed they could get a call to successfully go through from a cell phone to the toll free numbers, but not from land lines. After further testing it appears the some calls go through for most cell phone carriers and not others. It appears most land line calls to the toll free numbers fail, but there's been one or 2 land based offices that were able to successfully call the toll free numbers. All calls to the local numbers have went through without incident as to the best of our knowledge. We talked to the SIP provider and vendor for the SBC. They found that the SipXecs system is giving a "483 Too Many Hops" error. I've attached a capture (see the 726910-3.pcap file ) from the SIP provider as well as capture (See the "Configured by Ingate" file) from the SBC vendor that was ran on the SBC. You'll can view the capture from the SBC using a web browser, then scroll down toward the bottom to see the capture. You can search for the error code given above. The only changes we are aware of that happened on the SipXecs system, is that the selt signed cert had expired and was renewed about a month ago. The issue with the "483 Too Many Hops" has only been noticed by the client within the last 1.5 to 2 weeks. The toll free number of 8004144231 was used during the gathering of the 2 attached captures. Can someone please take a look at the attached capture files ASAP and let us know of any suggestions you have to resolve this issue? This issue is having a major impact the client's normal business operations. ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Upgrading sipx from 4.4 to 4.6
George you nailed it the backup from 4.4 did not have the .gz file extension. When I added the file extension leaving the rest of the file name intact it seems to have worked waiting for it to come back up now. Thank you very much! Jeff Deneau On Oct 19, 2012, at 10:28 AM, Tony Graziano wrote: > Your backup should be a tar.gz file. Are you unzipping it? > > If everything else is assumed the same (hostname, IP, password), the > procedure would be to make sure you sent the profile to the new server > once you log in before you do anything else. Then choose restore and > upload the file. If the file is really big, it might be desirable to > create a backup (onboard) the 4.6 system and replace the backup > file(s) in the backup directory with the ones from the 4.4 backup and > do a restore from "onboard". > > If you are getting error, look at the sipxconfig.og (in > /var/log/sipxpbx) for anything obvious and or post back with findings. > > On Fri, Oct 19, 2012 at 10:21 AM, Jeff Deneau wrote: >> I am running the current 4.4.0 release of SIPX (yum run just before backup) >> and just download 4.6.0 -- per the user forum I did a backup of 4.4.0 and >> loaded 4.6.0 on a new machine. >> Then attempted to do a restore from the file which was created on the 4.4 >> machine. >> I browse to the location pick the configuration.tar file and click restore I >> get a red message in the window that says >> [MESSAGE.WRONGFILETORESTORE] >> >> >> What is the proper process to upgrade from 4.4. to 4.6? >> >> ___ >> sipx-users mailing list >> sipx-users@list.sipfoundry.org >> List Archive: http://list.sipfoundry.org/archive/sipx-users/ > > > > -- > ~~ > Tony Graziano, Manager > Telephone: 434.984.8430 > sip: tgrazi...@voice.myitdepartment.net > Fax: 434.465.6833 > ~~ > Linked-In Profile: > http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 > Ask about our Internet Fax services! > ~~ > > Using or developing for sipXecs from SIPFoundry? Ask me about sipX-CoLab 2013! > > -- > LAN/Telephony/Security and Control Systems Helpdesk: > Telephone: 434.984.8426 > sip: helpd...@voice.myitdepartment.net > > Helpdesk Customers: http://myhelp.myitdepartment.net > Blog: http://blog.myitdepartment.net > ___ > sipx-users mailing list > sipx-users@list.sipfoundry.org > List Archive: http://list.sipfoundry.org/archive/sipx-users/ ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Getting 483 Too Many Hops Error
ingates are pretty resilient. the pcap is not attached. the cert if re-issues would have nothing to do with it. its more likely a problem with the dial plan within the ingate as it connects to broadvox. but as there is no pcap or details on this (some ITSP's use different dialplans for LD versus told free calls) but there is really not much information to go on. On Fri, Oct 19, 2012 at 1:31 PM, Brian Buckles wrote: > > > > > Sorry if this gets re-posted a second time. I haven't gotten any > confirmation from the admins that this was or wasn't accepted, and this is > something we need to resolve quickly. Please see my previous email below > for details. > Attention SipXecs support community, > We have a client that is running SipXecs 4.4.0. The Session Boarder > Controller is an InGate Siparator and the SIP provider for the SIP trunks is > BroadVox. The client is unable to receive calls to most of their toll free > numbers properly. Our client noticed that prior to calling us, that it > seemed they could get a call to successfully go through from a cell phone to > the toll free numbers, but not from land lines. After further testing it > appears the some calls go through for most cell phone carriers and not > others. It appears most land line calls to the toll free numbers fail, but > there's been one or 2 land based offices that were able to successfully call > the toll free numbers. All calls to the local numbers have went through > without incident as to the best of our knowledge. We talked to the SIP > provider and vendor for the SBC. They found that the SipXecs system is > giving a "483 Too Many Hops" error. I've attached a capture (see the > 726910-3.pcap file ) from the SIP provider as well as capture (See the > "Configured by Ingate" file) from the SBC vendor that was ran on the > SBC. You'll can view the capture from the SBC using a web browser, then > scroll down toward the bottom to see the capture. You can search for the > error code given above. The only changes we are aware of that happened on > the SipXecs system, is that the selt signed cert had expired and was renewed > about a month ago. The issue with the "483 Too Many Hops" has only been > noticed by the client within the last 1.5 to 2 weeks. The toll free number > of 8004144231 was used during the gathering of the 2 attached captures. Can > someone please take a look at the attached capture files ASAP and let us > know of any suggestions you have to resolve this issue? This issue is > having a major impact the client's normal business operations. > > > > ___ > sipx-users mailing list > sipx-users@list.sipfoundry.org > List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ Using or developing for sipXecs from SIPFoundry? Ask me about sipX-CoLab 2013! -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
[sipx-users] Getting 483 Too Many Hops Error
Sorry if this gets re-posted a second time. I haven't gotten any confirmation from the admins that this was or wasn't accepted, and this is something we need to resolve quickly. Please see my previous email below for details. Attention SipXecs support community, We have a client that is running SipXecs 4.4.0. The Session Boarder Controller is an InGate Siparator and the SIP provider for the SIP trunks is BroadVox. The client is unable to receive calls to most of their toll free numbers properly. Our client noticed that prior to calling us, that it seemed they could get a call to successfully go through from a cell phone to the toll free numbers, but not from land lines. After further testing it appears the some calls go through for most cell phone carriers and not others. It appears most land line calls to the toll free numbers fail, but there's been one or 2 land based offices that were able to successfully call the toll free numbers. All calls to the local numbers have went through without incident as to the best of our knowledge. We talked to the SIP provider and vendor for the SBC. They found that the SipXecs system is giving a "483 Too Many Hops" error. I've attached a capture (see the 726910-3.pcap file ) from the SIP provider as well as capture (See the "Configured by Ingate" file) from the SBC vendor that was ran on the SBC. You'll can view the capture from the SBC using a web browser, then scroll down toward the bottom to see the capture. You can search for the error code given above. The only changes we are aware of that happened on the SipXecs system, is that the selt signed cert had expired and was renewed about a month ago. The issue with the "483 Too Many Hops" has only been noticed by the client within the last 1.5 to 2 weeks. The toll free number of 8004144231 was used during the gathering of the 2 attached captures. Can someone please take a look at the attached capture files ASAP and let us know of any suggestions you have to resolve this issue? This issue is having a major impact the client's normal business operations.___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] New 4.6 Beta Installation
On Fri, Oct 19, 2012 at 12:22 PM, Aaron Pursell wrote: > cf3> -> Connect to 192.168.1.32 = 192.168.1.32 on port 5308 from 192.168.1.21 telnet 192.168.1.32 5308 if you cannot connect, run service stop iptables on all machines, then try again, if you still cannot connect, on 192.168.1.32 run /etc/init.d/sipxsupervisor stop /etc/init.d/sipxsupervisor nofork and see what console says when sending profiles ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] New 4.6 Beta Installation
Ran --reset-all on primary server and started over, re-added secondary to config page. Then ran --reset-all on secondary. No change. Aaron Pursell Network Security Administrator Easter Seals-Goodwill Northern Rocky Mountain, Inc. 4400 Central Ave Great Falls, Montana 59405 (406) 771-3721 aar...@esgw.org >>> "Aaron Pursell" 10/19/2012 10:22 AM >>> Ok did that and still fails but log shows: cf3> Initiate variable convergence... cf3> SET trustkey = 1 cf3> SET encrypt = 1 cf3> ... cf3> * Hailing 192.168.1.32 : 5308, with options "-v-K" (serial) cf3> ... cf3> No existing connection to 192.168.1.32 is established... cf3> Set cfengine port number to 5308 = 5308 cf3> Set connection timeout to 10 cf3> -> Connect to 192.168.1.32 = 192.168.1.32 on port 5308 cf3> !! Error connecting to server (timeout) cf3> !!! System error for connect: "Operation now in progress" cf3> !! Unable to connect to server 192.168.1.32 cf3> !!! System reports error for connect: "Operation now in progress" cf3> !! No server is responding on this port cf3> Unable to establish connection with 192.168.1.32 cf3> -> No suitable server responded to hail Aaron Pursell Network Security Administrator Easter Seals-Goodwill Northern Rocky Mountain, Inc. 4400 Central Ave Great Falls, Montana 59405 (406) 771-3721 aar...@esgw.org >>> George Niculae 10/19/2012 10:02 AM >>> On Fri, Oct 19, 2012 at 6:56 PM, Aaron Pursell wrote: The only service that runs on the .32 server is sipxsupervisor. No other services start...I'm new to 4.6 setup so if this is how it works then its fine. This was a fresh install but I re-ran sipxecs-setup with options --verbose and --reset. Can you try removing server from config, readding it and running sipxecs-setup --reset-all on slave? George ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] New 4.6 Beta Installation
Ok did that and still fails but log shows: cf3> Initiate variable convergence... cf3> SET trustkey = 1 cf3> SET encrypt = 1 cf3> ... cf3> * Hailing 192.168.1.32 : 5308, with options "-v-K" (serial) cf3> ... cf3> No existing connection to 192.168.1.32 is established... cf3> Set cfengine port number to 5308 = 5308 cf3> Set connection timeout to 10 cf3> -> Connect to 192.168.1.32 = 192.168.1.32 on port 5308 cf3> !! Error connecting to server (timeout) cf3> !!! System error for connect: "Operation now in progress" cf3> !! Unable to connect to server 192.168.1.32 cf3> !!! System reports error for connect: "Operation now in progress" cf3> !! No server is responding on this port cf3> Unable to establish connection with 192.168.1.32 cf3> -> No suitable server responded to hail Aaron Pursell Network Security Administrator Easter Seals-Goodwill Northern Rocky Mountain, Inc. 4400 Central Ave Great Falls, Montana 59405 (406) 771-3721 aar...@esgw.org >>> George Niculae 10/19/2012 10:02 AM >>> On Fri, Oct 19, 2012 at 6:56 PM, Aaron Pursell wrote: The only service that runs on the .32 server is sipxsupervisor. No other services start...I'm new to 4.6 setup so if this is how it works then its fine. This was a fresh install but I re-ran sipxecs-setup with options --verbose and --reset. Can you try removing server from config, readding it and running sipxecs-setup --reset-all on slave? George ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] New 4.6 Beta Installation
On Fri, Oct 19, 2012 at 6:56 PM, Aaron Pursell wrote: > The only service that runs on the .32 server is sipxsupervisor. No other > services start...I'm new to 4.6 setup so if this is how it works then its > fine. > > This was a fresh install but I re-ran sipxecs-setup with options --verbose > and --reset. > > Can you try removing server from config, readding it and running sipxecs-setup --reset-all on slave? George ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] New 4.6 Beta Installation
The only service that runs on the .32 server is sipxsupervisor. No other services start...I'm new to 4.6 setup so if this is how it works then its fine. This was a fresh install but I re-ran sipxecs-setup with options --verbose and --reset. I restarted the supervisor and on server 1 192.168.1.21 it lists the same error, "broken pipe". I found an old article relating to this exact error on 4.5.2 and I'm experiencing the same issues and Douglas Hubler stated then it was a bug and it should be resolved. There is no firewall or anything between server 1 .21 and now server 4 .32. .32 (secondary) shows its listening on 0.0.0.0:5308 and it has a connection established between 192.168.1.32:5308 and the first server on a high port. The Admin console still just reports: Configuration deployment10/19/12 9:51 AM10/19/12 9:51 AMFailedgfgwph.esgw.org:Error. System error for send: "Broken pipe". System error for send: "Broken pipe". System error for send: "Broken pipe". Authentication dialogue with 192.168.1.32 failed null Aaron Pursell Network Security Administrator Easter Seals-Goodwill Northern Rocky Mountain, Inc. 4400 Central Ave Great Falls, Montana 59405 (406) 771-3721 aar...@esgw.org >>> George Niculae 10/19/2012 9:31 AM >>> On Fri, Oct 19, 2012 at 6:18 PM, Aaron Pursell wrote: > On server 1 it reports: > > cf3> * Hailing 192.168.1.32 : 5308, with options "-v-K" (serial) > cf3> > ... > cf3> No existing connection to 192.168.1.32 is established... > cf3> Set cfengine port number to 5308 = 5308 > cf3> Set connection timeout to 10 > cf3> -> Connect to 192.168.1.32 = 192.168.1.32 on port 5308 > cf3> -> Did not find new key format /etc/sipxpbx/.cfagent/ppkeys/sipx-.pub > cf3> -> Trying old style /etc/sipxpbx/.cfagent/ppkeys/sipx-192.168.1.32.pub > cf3> Couldn't send > cf3> !!! System error for send: "Broken pipe" > cf3> Couldn't send > cf3> !!! System error for send: "Broken pipe" > cf3> Couldn't send > cf3> !!! System error for send: "Broken pipe" > cf3> Challenge response from server 192.168.1.32/192.168.1.32 was incorrect! > cf3> I: Report relates to a promise with handle "" > cf3> I: Promise is made internally by cfengine > cf3> !! Authentication dialogue with 192.168.1.32 failed > cf3> Unable to establish connection with 192.168.1.32 > cf3> -> No suitable server responded to hail > cf3> CFEngine - autonomous configuration engine - commence self-diagnostic > prelude > > On server 2 nothing ever hits the log. Is 192.168.1.32 your secondary IP? Can you try service sipxsupervisor restart on second machine and see if it works after (is the 2nd machine configured from the first shot, or have you run sipxecs-setup more than once) Thanks George ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] New 4.6 Beta Installation
On Fri, Oct 19, 2012 at 6:18 PM, Aaron Pursell wrote: > On server 1 it reports: > > cf3> * Hailing 192.168.1.32 : 5308, with options "-v-K" (serial) > cf3> > ... > cf3> No existing connection to 192.168.1.32 is established... > cf3> Set cfengine port number to 5308 = 5308 > cf3> Set connection timeout to 10 > cf3> -> Connect to 192.168.1.32 = 192.168.1.32 on port 5308 > cf3> -> Did not find new key format /etc/sipxpbx/.cfagent/ppkeys/sipx-.pub > cf3> -> Trying old style /etc/sipxpbx/.cfagent/ppkeys/sipx-192.168.1.32.pub > cf3> Couldn't send > cf3> !!! System error for send: "Broken pipe" > cf3> Couldn't send > cf3> !!! System error for send: "Broken pipe" > cf3> Couldn't send > cf3> !!! System error for send: "Broken pipe" > cf3> Challenge response from server 192.168.1.32/192.168.1.32 was incorrect! > cf3> I: Report relates to a promise with handle "" > cf3> I: Promise is made internally by cfengine > cf3> !! Authentication dialogue with 192.168.1.32 failed > cf3> Unable to establish connection with 192.168.1.32 > cf3> -> No suitable server responded to hail > cf3> CFEngine - autonomous configuration engine - commence self-diagnostic > prelude > > On server 2 nothing ever hits the log. Is 192.168.1.32 your secondary IP? Can you try service sipxsupervisor restart on second machine and see if it works after (is the 2nd machine configured from the first shot, or have you run sipxecs-setup more than once) Thanks George ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] New 4.6 Beta Installation
On server 1 it reports: cf3> * Hailing 192.168.1.32 : 5308, with options "-v-K" (serial) cf3> ... cf3> No existing connection to 192.168.1.32 is established... cf3> Set cfengine port number to 5308 = 5308 cf3> Set connection timeout to 10 cf3> -> Connect to 192.168.1.32 = 192.168.1.32 on port 5308 cf3> -> Did not find new key format /etc/sipxpbx/.cfagent/ppkeys/sipx-.pub cf3> -> Trying old style /etc/sipxpbx/.cfagent/ppkeys/sipx-192.168.1.32.pub cf3> Couldn't send cf3> !!! System error for send: "Broken pipe" cf3> Couldn't send cf3> !!! System error for send: "Broken pipe" cf3> Couldn't send cf3> !!! System error for send: "Broken pipe" cf3> Challenge response from server 192.168.1.32/192.168.1.32 was incorrect! cf3> I: Report relates to a promise with handle "" cf3> I: Promise is made internally by cfengine cf3> !! Authentication dialogue with 192.168.1.32 failed cf3> Unable to establish connection with 192.168.1.32 cf3> -> No suitable server responded to hail cf3> CFEngine - autonomous configuration engine - commence self-diagnostic prelude On server 2 nothing ever hits the log. Aaron Pursell Network Security Administrator Easter Seals-Goodwill Northern Rocky Mountain, Inc. 4400 Central Ave Great Falls, Montana 59405 (406) 771-3721 aar...@esgw.org >>> Mircea Carasel 10/19/2012 7:40 AM >>> Now when you make changes it just says, "System error for send: "Broken pipe". System error for send: "Broken pipe". Authentication dialogue with 192.168.1.32" What kind of changes are you making, please give us more details. From where did you got this error, is it from console? The only service running on server 2 is the cf engine nothing else. Anyone point me which log I can even look at or what I've done wrong? the cfengine gets logged into /var/log/sipxpbx/sipxagent.log Mircea Aaron Pursell Network Security Administrator Easter Seals-Goodwill Northern Rocky Mountain, Inc. 4400 Central Ave Great Falls, Montana 59405 (406) 771-3721 ( tel:%28406%29%20771-3721 ) aar...@esgw.org ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] New 4.6 Beta Installation
Anytime I make a user change it reports the config change was pushed to server 1 but then under failed jobs it says server 2 failed with the error below. I'm looking at the log now. Aaron Pursell Network Security Administrator Easter Seals-Goodwill Northern Rocky Mountain, Inc. 4400 Central Ave Great Falls, Montana 59405 (406) 771-3721 aar...@esgw.org >>> Mircea Carasel 10/19/2012 7:40 AM >>> Now when you make changes it just says, "System error for send: "Broken pipe". System error for send: "Broken pipe". Authentication dialogue with 192.168.1.32" What kind of changes are you making, please give us more details. From where did you got this error, is it from console? The only service running on server 2 is the cf engine nothing else. Anyone point me which log I can even look at or what I've done wrong? the cfengine gets logged into /var/log/sipxpbx/sipxagent.log Mircea Aaron Pursell Network Security Administrator Easter Seals-Goodwill Northern Rocky Mountain, Inc. 4400 Central Ave Great Falls, Montana 59405 (406) 771-3721 ( tel:%28406%29%20771-3721 ) aar...@esgw.org ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/ ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Upgrading sipx from 4.4 to 4.6
On Fri, Oct 19, 2012 at 5:28 PM, Tony Graziano wrote: > Your backup should be a tar.gz file. Are you unzipping it? > > If everything else is assumed the same (hostname, IP, password), the > procedure would be to make sure you sent the profile to the new server > once you log in before you do anything else. Then choose restore and > upload the file. If the file is really big, it might be desirable to > create a backup (onboard) the 4.6 system and replace the backup > file(s) in the backup directory with the ones from the 4.4 backup and > do a restore from "onboard". > > If you are getting error, look at the sipxconfig.og (in > /var/log/sipxpbx) for anything obvious and or post back with findings. > Also make sure that you upload configuration file in the right field within restore page (do not try to upload voicemail archive in config and viceversa) Thanks, George ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] Upgrading sipx from 4.4 to 4.6
Your backup should be a tar.gz file. Are you unzipping it? If everything else is assumed the same (hostname, IP, password), the procedure would be to make sure you sent the profile to the new server once you log in before you do anything else. Then choose restore and upload the file. If the file is really big, it might be desirable to create a backup (onboard) the 4.6 system and replace the backup file(s) in the backup directory with the ones from the 4.4 backup and do a restore from "onboard". If you are getting error, look at the sipxconfig.og (in /var/log/sipxpbx) for anything obvious and or post back with findings. On Fri, Oct 19, 2012 at 10:21 AM, Jeff Deneau wrote: > I am running the current 4.4.0 release of SIPX (yum run just before backup) > and just download 4.6.0 -- per the user forum I did a backup of 4.4.0 and > loaded 4.6.0 on a new machine. > Then attempted to do a restore from the file which was created on the 4.4 > machine. > I browse to the location pick the configuration.tar file and click restore I > get a red message in the window that says > [MESSAGE.WRONGFILETORESTORE] > > > What is the proper process to upgrade from 4.4. to 4.6? > > ___ > sipx-users mailing list > sipx-users@list.sipfoundry.org > List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ Using or developing for sipXecs from SIPFoundry? Ask me about sipX-CoLab 2013! -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: helpd...@voice.myitdepartment.net Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
[sipx-users] Upgrading sipx from 4.4 to 4.6
I am running the current 4.4.0 release of SIPX (yum run just before backup) and just download 4.6.0 -- per the user forum I did a backup of 4.4.0 and loaded 4.6.0 on a new machine. Then attempted to do a restore from the file which was created on the 4.4 machine. I browse to the location pick the configuration.tar file and click restore I get a red message in the window that says [MESSAGE.WRONGFILETORESTORE] What is the proper process to upgrade from 4.4. to 4.6?___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] openacd 4.6 connections refused
Good to know. Thank you! From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Laurentiu Ceausescu Sent: Friday, October 19, 2012 9:09 AM To: Discussion list for users of sipXecs software Subject: Re: [sipx-users] openacd 4.6 connections refused On Fri, Oct 19, 2012 at 5:04 PM, Geoff Musgrave mailto:geoff.musgr...@cacionline.net>> wrote: False alarm. Somehow my edits to iptables were lost between yesterday and today and the default iptables entries were blocking access to port 5050 for the agent UI. With XX-10507 (committed yesterday) you can allow the openacd web UI port from System/Firewall page Laurentiu ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] openacd 4.6 connections refused
On Fri, Oct 19, 2012 at 5:04 PM, Geoff Musgrave < geoff.musgr...@cacionline.net> wrote: > False alarm. > > ** ** > > Somehow my edits to iptables were lost between yesterday and today and the > default iptables entries were blocking access to port 5050 for the agent UI. > With XX-10507 (committed yesterday) you can allow the openacd web UI port from System/Firewall page Laurentiu ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] openacd 4.6 connections refused
False alarm. Somehow my edits to iptables were lost between yesterday and today and the default iptables entries were blocking access to port 5050 for the agent UI. Thank you. From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Geoff Musgrave Sent: Friday, October 19, 2012 8:40 AM To: Discussion list for users of sipXecs software (sipx-users@list.sipfoundry.org) Subject: [sipx-users] openacd 4.6 connections refused Neither the admin or the agent UI's will load this morning for openacd. They both worked yesterday and yesterday evening and no updates were done to the sipXecs server last night. I've restarted services and rebooted the server but still the problem remains. I've attached the full.log file for openacd which shows connections being refused. I'm working on this now and any input or direction is appreciated. Version: sipXecs (4.6.0. 2012-10-02EDT16:10:02 ip-10-72-43-110.ec2.internal) update unknown modified Thank you ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] New 4.6 Beta Installation
> > > > Now when you make changes it just says, "System error for send: "Broken > pipe". System error for send: "Broken pipe". Authentication dialogue with > 192.168.1.32" > What kind of changes are you making, please give us more details. From where did you got this error, is it from console? > > The only service running on server 2 is the cf engine nothing else. Anyone > point me which log I can even look at or what I've done wrong? > the cfengine gets logged into /var/log/sipxpbx/sipxagent.log Mircea > > > Aaron Pursell > Network Security Administrator > Easter Seals-Goodwill Northern Rocky Mountain, Inc. > 4400 Central Ave > Great Falls, Montana 59405 > > (406) 771-3721 > aar...@esgw.org > > ___ > sipx-users mailing list > sipx-users@list.sipfoundry.org > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
[sipx-users] New 4.6 Beta Installation
So, I've been trying to work on this new system and I've probably missed something or done something minute to create this issue however I have a master server 1 at 192.168.1.21 and a "slave" at 192.168.1.32 as server 2. When I setup server 2 I entered the correct number assigned and it moved from not configured to configured. Now when you make changes it just says, "System error for send: "Broken pipe". System error for send: "Broken pipe". Authentication dialogue with 192.168.1.32" The only service running on server 2 is the cf engine nothing else. Anyone point me which log I can even look at or what I've done wrong? Aaron Pursell Network Security Administrator Easter Seals-Goodwill Northern Rocky Mountain, Inc. 4400 Central Ave Great Falls, Montana 59405 (406) 771-3721 aar...@esgw.org ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] 4.4.0 to 4.4.6
On Fri, Oct 19, 2012 at 1:22 PM, George Niculae wrote: > On Fri, Oct 19, 2012 at 4:50 AM, Andrew Pitman > wrote: >> Thanks George! I tried it again today and verified it works. It seems that >> the heard/unheard status of voicemails is not preserved though. All >> messages were "new" after the import. It's no biggie, but is this something >> that is being worked on or something we'll just have to deal with? > > Good catch, let me fix that too. Thanks! > Fix committed, if you're installing from openuc-stage then yum update, if you're installing from openuc then it will take some more time until build published. Either way please let me know if works for you after yum update and test it, I have a button to push :) George ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] 4.4.0 to 4.4.6
On Fri, Oct 19, 2012 at 4:50 AM, Andrew Pitman wrote: > Thanks George! I tried it again today and verified it works. It seems that > the heard/unheard status of voicemails is not preserved though. All messages > were "new" after the import. It's no biggie, but is this something that is > being worked on or something we'll just have to deal with? Good catch, let me fix that too. Thanks! George ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
Re: [sipx-users] External calls cannot be transferred to voice mail (sipXecs 4.4.0)
Primus is also LINGO. Primus is a large aggregator and also runs a residential service (Lingo). The Lingo service does not support invite without sdp, unless the reinvite is to one of their services and typically only from one of their ATA's. I think you would do well to ask them if they support this AND it is very important to make sure the invite for the incoming call comes to your server on port 5080. I don't think your issue is unusual and usually stems from one of 3 core misconfiguration types: 1. Incompatible ITSP - Does a transfer from the AA to a user work? Does a call from a user to another user work? (both as inbound calls via the trunk). Is the original invite coming on port 5080. 2. Does the phone ring? If so, how was it configured (manually of by sipx)? Please tell me you didn't register the line manually using the sipx ip address. DNS is VERY important for the refer to voicemail. IF you registered by IP, make sure you add the IP as a domain alias, but really you should NEVER register by IP and expect all things to work well. 3. Firewall - Unless your firewall is doing 1:1 NAT, you need to make sure it can do Manual AON (static port nat). With pfsense this is easy, but YOU CANNOT create port forward rules until you do this for SIPX becuase they will follow the original NAT type. I sent you a link of how to do this earlier, its pretty straightforward. You should be able to use the pcap tool in pfsense and have it listen on WAN port 5080 and do a capture and see if the ITSP sends the call in on the right port. If not, it will never work right (no matter what version) and you need to ask them if they support this. Good luck! On Fri, Oct 19, 2012 at 1:50 AM, Joegen Baclor wrote: > Transferring ITSP originated calls requires that your ITSP supports INVITE > without SDP. Before barking on something on the system, check first if your > ITSP supports this. If not, there is no way your ITSP will work with sipx > initiated transfers. > > > > On 10/19/2012 01:19 PM, Tony Graziano wrote: >> >> On Fri, Oct 19, 2012 at 1:07 AM, Henry Kwan wrote: >>> >>> My installation was right from the 4.4 ISO. I did try without updating >>> at >>> all but to no avail. >>> >>> My ITSP is Primus Canada. >>> >>> Well I have to admit that I am not knowledgeable in setting up pfSense. >>> In >>> fact I am not knowledgeable on how to produce a pcap or produce a >>> siptrace >>> as Tony suggested. Having said that, I'll continue to play with 4.4 and >>> look into how to perform the tasks suggested when time permits. >>> >> Pfsense >> http://blog.myitdepartment.net/?p=297 >>> >>> In the mean time, 4.2.1 will have to suffice until I can figure out what >>> I >>> did wrong. >>> >>> By the way, my observation regarding the inconsistent behaviour on >>> restarts >>> for both 4.2.1 and 4.4.0 is not an attack on sipXecs. It is just that - >>> an >>> observation. Maybe someone can comment if this observation is also only >>> experienced by me. If that's the case, I must be a jinx or have a unique >>> ability to bring out the worst in sipXecs. >>> >> I can set up a new system each day and don't experience this behavior. >> It's really important to observe how much RAM you have installed (I >> prefer 8GB minimum but for very small installs 4GB on 4.4 is enough, >> though 8GB should be the minimum for 4.6). >>> >>> Best regards to all, >>> >>> Henry Kwan >>> >>> >>> From: George Niculae >>> >>> To: Discussion list for users of sipXecs software >>> >>> Cc: Henry Kwan >>> Sent: Thursday, October 18, 2012 6:29:01 PM >>> >>> Subject: Re: [sipx-users] External calls cannot be transferred to voice >>> mail >>> (sipXecs 4.4.0) >>> >>> On Fri, Oct 19, 2012 at 3:26 AM, Tony Graziano >>> wrote: Rather than use an old unsupportable version, produce a pcap from your firewall or produce a siptrace from sipx itself. I don't think your off the cuff observation is exactly right on targetm . Version 4.2 used its own media server while 4.4 uses FreeSWITCH and there are significant close changes. You could also indicate whether or not you followed a tutorial on how to properly configure pfsense and who the itsp is. >>> Additionally, if you could try scenario with 4.4 built from ISO, >>> without yum updating to latest, and report back, will help identifying >>> if issue in latest patches >>> >>> Thanks >>> George >>> >>> >>> >>> ___ >>> sipx-users mailing list >>> sipx-users@list.sipfoundry.org >>> List Archive: http://list.sipfoundry.org/archive/sipx-users/ >> >> >> > -- ~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: tgrazi...@voice.myitdepartment.net Fax: 434.465.6833 ~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~ Using or developing for sipXecs from SIPFoundry? Ask me
Re: [sipx-users] Unable to download list of phones/users using SOAP
On Fri, Oct 19, 2012 at 9:36 AM, Russell Fox wrote:
>
>
> I’m trying to get a list of phones (users too) using SOAP, but get no
> records returned even though I use the exact serialnumber that I added the
> record with.
>
Hm, I just double checked on a 4.4 machine using SoapUI and I cannot
get response back either, I am looking into and will come back
>
>
> I’ve tried this against version 4.4 and 4.6 (originally thought it was a
> problem on 4.4). The code adds the record successfully (I’ve checked via the
> web site) but fails to retrieve it back again. I’ve also tried with an empty
> value in search.bySerialNumber and by setting the group and searching on
> that.
>
>
>
> I’d prefer to be doing these calls via REST services, but couldn’t find the
> relevant documentation for Phones and Users for REST.
>
Unfortunately there is no REST API at this moment for this (though it
should be pretty easy to add)
George
class Program
{
static void Main(string[] args)
{
NetworkCredential sipXCredential = new
NetworkCredential(SipXUserName, SipXPassword);
//Disable problem with invalid certificate on the site
// - should probably be removed later on, but keep while figuring
this out.
ServicePointManager.ServerCertificateValidationCallback =
delegate(object s, X509Certificate certificate, X509Chain chain,
SslPolicyErrors sslPolicyErrors) { return true; };
PhoneService4_6 phone_service = new PhoneService4_6();
phone_service.Credentials = sipXCredential;
Phone seed = new Phone() { serialNumber = "0001", modelId =
"acmePhoneStandard" };
AddPhone addPhone = new AddPhone() { phone = seed };
try
{
Console.WriteLine(string.Format("Adding Phone: {0}",
addPhone.phone.serialNumber));
phone_service.addPhone(addPhone);
Console.WriteLine("Added phone");
}
catch (Exception e)
{
Console.WriteLine(string.Format("Error: {0}", e.Message));
}
Console.WriteLine();
PhoneSearch search = new PhoneSearch() { bySerialNumber =
seed.serialNumber};
try
{
Console.WriteLine(string.Format("findPhone: {0}",
search.bySerialNumber));
var all = phone_service.findPhone(new FindPhone() { search =
search }).phones;
if (all.Count() != 0)
{
foreach (var phone in all)
{
Console.WriteLine(string.Format("Phone returned: {0}",
phone.serialNumber));
}
}
Console.WriteLine(string.Format("findPhone returned {0}",
all.Count().ToString()));
}
catch (Exception e)
{
Console.WriteLine(string.Format("Error: {0}", e.Message));
}
Console.ReadLine();
}
}___
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